A Voice Over IP (VoIP) call could be directed along a different media path, such as a Public Switched Telephone Network (PSTN), when the IP network carrying the VoIP call is suffering from poor quality. Gateways in the call path could also be notified to use different COmpressor/DECompressors (codecs), etc. when low quality of service is detected in the IP network. A billing system could bill a subscriber at a lower rate when the quality of a call is compromised. Real-time billing data could also be derived if a system could determine the number of bits transmitted or the type of media services used during a packet switched network connection. For example, different billing tables could be used for voice, fax, modem relay, or pass through calls.
Accordingly, different network processing devices, such as call agents, Session Initiated Protocol (SIP) proxies and H.323 gatekeepers, have an interest in monitoring media liveness and heath conditions. H.323 is a standard approved by the International Telecommunication Union (ITU) that defines how audiovisual data is transmitted across networks.
Media including audio and video data is transported over a packet switched network using a Real-time Transport Protocol (RTP) protocol. A Real-time Control Protocol (RTCP) is then used to report statistical information regarding the RTP connection. The RTCP reports contain information regarding the number of packets transmitted, the number of packets lost, latency, jitter information, etc. for IP packets in the RTP connection.
The problem is that the RTCP reports are only sent between the two gateways in the packet network that are conducting the RTP session and are not available to other network elements that may want to use the reports for policy or billing reasons.
The present invention addresses this and other problems associated with the prior art.